Libav
aacenc.c
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1 /*
2  * AAC encoder
3  * Copyright (C) 2008 Konstantin Shishkov
4  *
5  * This file is part of Libav.
6  *
7  * Libav is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * Libav is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
27 /***********************************
28  * TODOs:
29  * add sane pulse detection
30  * add temporal noise shaping
31  ***********************************/
32 
33 #include "libavutil/float_dsp.h"
34 #include "libavutil/opt.h"
35 #include "avcodec.h"
36 #include "put_bits.h"
37 #include "internal.h"
38 #include "mpeg4audio.h"
39 #include "kbdwin.h"
40 #include "sinewin.h"
41 
42 #include "aac.h"
43 #include "aactab.h"
44 #include "aacenc.h"
45 
46 #include "psymodel.h"
47 
48 #define AAC_MAX_CHANNELS 6
49 
50 #define ERROR_IF(cond, ...) \
51  if (cond) { \
52  av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
53  return AVERROR(EINVAL); \
54  }
55 
56 float ff_aac_pow34sf_tab[428];
57 
58 static const uint8_t swb_size_1024_96[] = {
59  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
60  12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
61  64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
62 };
63 
64 static const uint8_t swb_size_1024_64[] = {
65  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
66  12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
67  40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
68 };
69 
70 static const uint8_t swb_size_1024_48[] = {
71  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
72  12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
73  32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
74  96
75 };
76 
77 static const uint8_t swb_size_1024_32[] = {
78  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
79  12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
80  32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
81 };
82 
83 static const uint8_t swb_size_1024_24[] = {
84  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
85  12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
86  32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
87 };
88 
89 static const uint8_t swb_size_1024_16[] = {
90  8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
91  12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
92  32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
93 };
94 
95 static const uint8_t swb_size_1024_8[] = {
96  12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
97  16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
98  32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
99 };
100 
101 static const uint8_t *swb_size_1024[] = {
106 };
107 
108 static const uint8_t swb_size_128_96[] = {
109  4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
110 };
111 
112 static const uint8_t swb_size_128_48[] = {
113  4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
114 };
115 
116 static const uint8_t swb_size_128_24[] = {
117  4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
118 };
119 
120 static const uint8_t swb_size_128_16[] = {
121  4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
122 };
123 
124 static const uint8_t swb_size_128_8[] = {
125  4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
126 };
127 
128 static const uint8_t *swb_size_128[] = {
129  /* the last entry on the following row is swb_size_128_64 but is a
130  duplicate of swb_size_128_96 */
135 };
136 
138 static const uint8_t aac_chan_configs[6][5] = {
139  {1, TYPE_SCE}, // 1 channel - single channel element
140  {1, TYPE_CPE}, // 2 channels - channel pair
141  {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
142  {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
143  {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
144  {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
145 };
146 
151  { 0 },
152  { 0, 1 },
153  { 2, 0, 1 },
154  { 2, 0, 1, 3 },
155  { 2, 0, 1, 3, 4 },
156  { 2, 0, 1, 4, 5, 3 },
157 };
158 
164 {
165  PutBitContext pb;
166  AACEncContext *s = avctx->priv_data;
167 
168  init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
169  put_bits(&pb, 5, 2); //object type - AAC-LC
170  put_bits(&pb, 4, s->samplerate_index); //sample rate index
171  put_bits(&pb, 4, s->channels);
172  //GASpecificConfig
173  put_bits(&pb, 1, 0); //frame length - 1024 samples
174  put_bits(&pb, 1, 0); //does not depend on core coder
175  put_bits(&pb, 1, 0); //is not extension
176 
177  //Explicitly Mark SBR absent
178  put_bits(&pb, 11, 0x2b7); //sync extension
179  put_bits(&pb, 5, AOT_SBR);
180  put_bits(&pb, 1, 0);
181  flush_put_bits(&pb);
182 }
183 
184 #define WINDOW_FUNC(type) \
185 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
186  SingleChannelElement *sce, \
187  const float *audio)
188 
189 WINDOW_FUNC(only_long)
190 {
191  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
192  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
193  float *out = sce->ret_buf;
194 
195  fdsp->vector_fmul (out, audio, lwindow, 1024);
196  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
197 }
198 
199 WINDOW_FUNC(long_start)
200 {
201  const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
202  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
203  float *out = sce->ret_buf;
204 
205  fdsp->vector_fmul(out, audio, lwindow, 1024);
206  memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
207  fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
208  memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
209 }
210 
211 WINDOW_FUNC(long_stop)
212 {
213  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
214  const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
215  float *out = sce->ret_buf;
216 
217  memset(out, 0, sizeof(out[0]) * 448);
218  fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
219  memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
220  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
221 }
222 
223 WINDOW_FUNC(eight_short)
224 {
225  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
226  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
227  const float *in = audio + 448;
228  float *out = sce->ret_buf;
229  int w;
230 
231  for (w = 0; w < 8; w++) {
232  fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
233  out += 128;
234  in += 128;
235  fdsp->vector_fmul_reverse(out, in, swindow, 128);
236  out += 128;
237  }
238 }
239 
240 static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
242  const float *audio) = {
243  [ONLY_LONG_SEQUENCE] = apply_only_long_window,
244  [LONG_START_SEQUENCE] = apply_long_start_window,
245  [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
246  [LONG_STOP_SEQUENCE] = apply_long_stop_window
247 };
248 
250  float *audio)
251 {
252  int i;
253  float *output = sce->ret_buf;
254 
255  apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio);
256 
258  s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
259  else
260  for (i = 0; i < 1024; i += 128)
261  s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
262  memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
263 }
264 
270 {
271  int w;
272 
273  put_bits(&s->pb, 1, 0); // ics_reserved bit
274  put_bits(&s->pb, 2, info->window_sequence[0]);
275  put_bits(&s->pb, 1, info->use_kb_window[0]);
276  if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
277  put_bits(&s->pb, 6, info->max_sfb);
278  put_bits(&s->pb, 1, 0); // no prediction
279  } else {
280  put_bits(&s->pb, 4, info->max_sfb);
281  for (w = 1; w < 8; w++)
282  put_bits(&s->pb, 1, !info->group_len[w]);
283  }
284 }
285 
291 {
292  int i, w;
293 
294  put_bits(pb, 2, cpe->ms_mode);
295  if (cpe->ms_mode == 1)
296  for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
297  for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
298  put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
299 }
300 
304 static void adjust_frame_information(ChannelElement *cpe, int chans)
305 {
306  int i, w, w2, g, ch;
307  int start, maxsfb, cmaxsfb;
308 
309  for (ch = 0; ch < chans; ch++) {
310  IndividualChannelStream *ics = &cpe->ch[ch].ics;
311  start = 0;
312  maxsfb = 0;
313  cpe->ch[ch].pulse.num_pulse = 0;
314  for (w = 0; w < ics->num_windows*16; w += 16) {
315  for (g = 0; g < ics->num_swb; g++) {
316  //apply M/S
317  if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {
318  for (i = 0; i < ics->swb_sizes[g]; i++) {
319  cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
320  cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
321  }
322  }
323  start += ics->swb_sizes[g];
324  }
325  for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
326  ;
327  maxsfb = FFMAX(maxsfb, cmaxsfb);
328  }
329  ics->max_sfb = maxsfb;
330 
331  //adjust zero bands for window groups
332  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
333  for (g = 0; g < ics->max_sfb; g++) {
334  i = 1;
335  for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
336  if (!cpe->ch[ch].zeroes[w2*16 + g]) {
337  i = 0;
338  break;
339  }
340  }
341  cpe->ch[ch].zeroes[w*16 + g] = i;
342  }
343  }
344  }
345 
346  if (chans > 1 && cpe->common_window) {
347  IndividualChannelStream *ics0 = &cpe->ch[0].ics;
348  IndividualChannelStream *ics1 = &cpe->ch[1].ics;
349  int msc = 0;
350  ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
351  ics1->max_sfb = ics0->max_sfb;
352  for (w = 0; w < ics0->num_windows*16; w += 16)
353  for (i = 0; i < ics0->max_sfb; i++)
354  if (cpe->ms_mask[w+i])
355  msc++;
356  if (msc == 0 || ics0->max_sfb == 0)
357  cpe->ms_mode = 0;
358  else
359  cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
360  }
361 }
362 
367 {
368  int w;
369 
370  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
371  s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
372 }
373 
379 {
380  int off = sce->sf_idx[0], diff;
381  int i, w;
382 
383  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
384  for (i = 0; i < sce->ics.max_sfb; i++) {
385  if (!sce->zeroes[w*16 + i]) {
386  diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
387  if (diff < 0 || diff > 120)
388  av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
389  off = sce->sf_idx[w*16 + i];
391  }
392  }
393  }
394 }
395 
399 static void encode_pulses(AACEncContext *s, Pulse *pulse)
400 {
401  int i;
402 
403  put_bits(&s->pb, 1, !!pulse->num_pulse);
404  if (!pulse->num_pulse)
405  return;
406 
407  put_bits(&s->pb, 2, pulse->num_pulse - 1);
408  put_bits(&s->pb, 6, pulse->start);
409  for (i = 0; i < pulse->num_pulse; i++) {
410  put_bits(&s->pb, 5, pulse->pos[i]);
411  put_bits(&s->pb, 4, pulse->amp[i]);
412  }
413 }
414 
419 {
420  int start, i, w, w2;
421 
422  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
423  start = 0;
424  for (i = 0; i < sce->ics.max_sfb; i++) {
425  if (sce->zeroes[w*16 + i]) {
426  start += sce->ics.swb_sizes[i];
427  continue;
428  }
429  for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
430  s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
431  sce->ics.swb_sizes[i],
432  sce->sf_idx[w*16 + i],
433  sce->band_type[w*16 + i],
434  s->lambda);
435  start += sce->ics.swb_sizes[i];
436  }
437  }
438 }
439 
445  int common_window)
446 {
447  put_bits(&s->pb, 8, sce->sf_idx[0]);
448  if (!common_window)
449  put_ics_info(s, &sce->ics);
450  encode_band_info(s, sce);
451  encode_scale_factors(avctx, s, sce);
452  encode_pulses(s, &sce->pulse);
453  put_bits(&s->pb, 1, 0); //tns
454  put_bits(&s->pb, 1, 0); //ssr
455  encode_spectral_coeffs(s, sce);
456  return 0;
457 }
458 
462 static void put_bitstream_info(AACEncContext *s, const char *name)
463 {
464  int i, namelen, padbits;
465 
466  namelen = strlen(name) + 2;
467  put_bits(&s->pb, 3, TYPE_FIL);
468  put_bits(&s->pb, 4, FFMIN(namelen, 15));
469  if (namelen >= 15)
470  put_bits(&s->pb, 8, namelen - 14);
471  put_bits(&s->pb, 4, 0); //extension type - filler
472  padbits = -put_bits_count(&s->pb) & 7;
474  for (i = 0; i < namelen - 2; i++)
475  put_bits(&s->pb, 8, name[i]);
476  put_bits(&s->pb, 12 - padbits, 0);
477 }
478 
479 /*
480  * Copy input samples.
481  * Channels are reordered from Libav's default order to AAC order.
482  */
483 static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
484 {
485  int ch;
486  int end = 2048 + (frame ? frame->nb_samples : 0);
487  const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
488 
489  /* copy and remap input samples */
490  for (ch = 0; ch < s->channels; ch++) {
491  /* copy last 1024 samples of previous frame to the start of the current frame */
492  memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
493 
494  /* copy new samples and zero any remaining samples */
495  if (frame) {
496  memcpy(&s->planar_samples[ch][2048],
497  frame->extended_data[channel_map[ch]],
498  frame->nb_samples * sizeof(s->planar_samples[0][0]));
499  }
500  memset(&s->planar_samples[ch][end], 0,
501  (3072 - end) * sizeof(s->planar_samples[0][0]));
502  }
503 }
504 
505 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
506  const AVFrame *frame, int *got_packet_ptr)
507 {
508  AACEncContext *s = avctx->priv_data;
509  float **samples = s->planar_samples, *samples2, *la, *overlap;
510  ChannelElement *cpe;
511  int i, ch, w, g, chans, tag, start_ch, ret;
512  int chan_el_counter[4];
514 
515  if (s->last_frame == 2)
516  return 0;
517 
518  /* add current frame to queue */
519  if (frame) {
520  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
521  return ret;
522  }
523 
524  copy_input_samples(s, frame);
525  if (s->psypp)
527 
528  if (!avctx->frame_number)
529  return 0;
530 
531  start_ch = 0;
532  for (i = 0; i < s->chan_map[0]; i++) {
533  FFPsyWindowInfo* wi = windows + start_ch;
534  tag = s->chan_map[i+1];
535  chans = tag == TYPE_CPE ? 2 : 1;
536  cpe = &s->cpe[i];
537  for (ch = 0; ch < chans; ch++) {
538  IndividualChannelStream *ics = &cpe->ch[ch].ics;
539  int cur_channel = start_ch + ch;
540  overlap = &samples[cur_channel][0];
541  samples2 = overlap + 1024;
542  la = samples2 + (448+64);
543  if (!frame)
544  la = NULL;
545  if (tag == TYPE_LFE) {
546  wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
547  wi[ch].window_shape = 0;
548  wi[ch].num_windows = 1;
549  wi[ch].grouping[0] = 1;
550 
551  /* Only the lowest 12 coefficients are used in a LFE channel.
552  * The expression below results in only the bottom 8 coefficients
553  * being used for 11.025kHz to 16kHz sample rates.
554  */
555  ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
556  } else {
557  wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
558  ics->window_sequence[0]);
559  }
560  ics->window_sequence[1] = ics->window_sequence[0];
561  ics->window_sequence[0] = wi[ch].window_type[0];
562  ics->use_kb_window[1] = ics->use_kb_window[0];
563  ics->use_kb_window[0] = wi[ch].window_shape;
564  ics->num_windows = wi[ch].num_windows;
565  ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
566  ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
567  for (w = 0; w < ics->num_windows; w++)
568  ics->group_len[w] = wi[ch].grouping[w];
569 
570  apply_window_and_mdct(s, &cpe->ch[ch], overlap);
571  }
572  start_ch += chans;
573  }
574  if ((ret = ff_alloc_packet(avpkt, 768 * s->channels))) {
575  av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
576  return ret;
577  }
578 
579  do {
580  int frame_bits;
581 
582  init_put_bits(&s->pb, avpkt->data, avpkt->size);
583 
584  if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
586  start_ch = 0;
587  memset(chan_el_counter, 0, sizeof(chan_el_counter));
588  for (i = 0; i < s->chan_map[0]; i++) {
589  FFPsyWindowInfo* wi = windows + start_ch;
590  const float *coeffs[2];
591  tag = s->chan_map[i+1];
592  chans = tag == TYPE_CPE ? 2 : 1;
593  cpe = &s->cpe[i];
594  put_bits(&s->pb, 3, tag);
595  put_bits(&s->pb, 4, chan_el_counter[tag]++);
596  for (ch = 0; ch < chans; ch++)
597  coeffs[ch] = cpe->ch[ch].coeffs;
598  s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
599  for (ch = 0; ch < chans; ch++) {
600  s->cur_channel = start_ch + ch;
601  s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
602  }
603  cpe->common_window = 0;
604  if (chans > 1
605  && wi[0].window_type[0] == wi[1].window_type[0]
606  && wi[0].window_shape == wi[1].window_shape) {
607 
608  cpe->common_window = 1;
609  for (w = 0; w < wi[0].num_windows; w++) {
610  if (wi[0].grouping[w] != wi[1].grouping[w]) {
611  cpe->common_window = 0;
612  break;
613  }
614  }
615  }
616  s->cur_channel = start_ch;
617  if (s->options.stereo_mode && cpe->common_window) {
618  if (s->options.stereo_mode > 0) {
619  IndividualChannelStream *ics = &cpe->ch[0].ics;
620  for (w = 0; w < ics->num_windows; w += ics->group_len[w])
621  for (g = 0; g < ics->num_swb; g++)
622  cpe->ms_mask[w*16+g] = 1;
623  } else if (s->coder->search_for_ms) {
624  s->coder->search_for_ms(s, cpe, s->lambda);
625  }
626  }
627  adjust_frame_information(cpe, chans);
628  if (chans == 2) {
629  put_bits(&s->pb, 1, cpe->common_window);
630  if (cpe->common_window) {
631  put_ics_info(s, &cpe->ch[0].ics);
632  encode_ms_info(&s->pb, cpe);
633  }
634  }
635  for (ch = 0; ch < chans; ch++) {
636  s->cur_channel = start_ch + ch;
637  encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
638  }
639  start_ch += chans;
640  }
641 
642  frame_bits = put_bits_count(&s->pb);
643  if (frame_bits <= 6144 * s->channels - 3) {
644  s->psy.bitres.bits = frame_bits / s->channels;
645  break;
646  }
647 
648  s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
649 
650  } while (1);
651 
652  put_bits(&s->pb, 3, TYPE_END);
653  flush_put_bits(&s->pb);
654  avctx->frame_bits = put_bits_count(&s->pb);
655 
656  // rate control stuff
657  if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
658  float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
659  s->lambda *= ratio;
660  s->lambda = FFMIN(s->lambda, 65536.f);
661  }
662 
663  if (!frame)
664  s->last_frame++;
665 
666  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
667  &avpkt->duration);
668 
669  avpkt->size = put_bits_count(&s->pb) >> 3;
670  *got_packet_ptr = 1;
671  return 0;
672 }
673 
675 {
676  AACEncContext *s = avctx->priv_data;
677 
678  ff_mdct_end(&s->mdct1024);
679  ff_mdct_end(&s->mdct128);
680  ff_psy_end(&s->psy);
681  if (s->psypp)
683  av_freep(&s->buffer.samples);
684  av_freep(&s->cpe);
685  ff_af_queue_close(&s->afq);
686  return 0;
687 }
688 
690 {
691  int ret = 0;
692 
694 
695  // window init
700 
701  if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
702  return ret;
703  if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0))
704  return ret;
705 
706  return 0;
707 }
708 
710 {
711  int ch;
712  FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
713  FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
714  FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
715 
716  for(ch = 0; ch < s->channels; ch++)
717  s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
718 
719  return 0;
720 alloc_fail:
721  return AVERROR(ENOMEM);
722 }
723 
725 {
726  AACEncContext *s = avctx->priv_data;
727  int i, ret = 0;
728  const uint8_t *sizes[2];
729  uint8_t grouping[AAC_MAX_CHANNELS];
730  int lengths[2];
731 
732  avctx->frame_size = 1024;
733 
734  for (i = 0; i < 16; i++)
736  break;
737 
738  s->channels = avctx->channels;
739 
740  ERROR_IF(i == 16,
741  "Unsupported sample rate %d\n", avctx->sample_rate);
743  "Unsupported number of channels: %d\n", s->channels);
745  "Unsupported profile %d\n", avctx->profile);
746  ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
747  "Too many bits per frame requested\n");
748 
749  s->samplerate_index = i;
750 
752 
753  if ((ret = dsp_init(avctx, s)) < 0)
754  goto fail;
755 
756  if ((ret = alloc_buffers(avctx, s)) < 0)
757  goto fail;
758 
759  avctx->extradata_size = 5;
761 
762  sizes[0] = swb_size_1024[i];
763  sizes[1] = swb_size_128[i];
764  lengths[0] = ff_aac_num_swb_1024[i];
765  lengths[1] = ff_aac_num_swb_128[i];
766  for (i = 0; i < s->chan_map[0]; i++)
767  grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
768  if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
769  s->chan_map[0], grouping)) < 0)
770  goto fail;
771  s->psypp = ff_psy_preprocess_init(avctx);
772  s->coder = &ff_aac_coders[2];
773 
774  s->lambda = avctx->global_quality ? avctx->global_quality : 120;
775 
777 
778  for (i = 0; i < 428; i++)
779  ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
780 
781  avctx->delay = 1024;
782  ff_af_queue_init(avctx, &s->afq);
783 
784  return 0;
785 fail:
786  aac_encode_end(avctx);
787  return ret;
788 }
789 
790 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
791 static const AVOption aacenc_options[] = {
792  {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
793  {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
794  {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
795  {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
796  {NULL}
797 };
798 
799 static const AVClass aacenc_class = {
800  "AAC encoder",
804 };
805 
807  .name = "aac",
808  .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
809  .type = AVMEDIA_TYPE_AUDIO,
810  .id = AV_CODEC_ID_AAC,
811  .priv_data_size = sizeof(AACEncContext),
813  .encode2 = aac_encode_frame,
815  .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
817  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
819  .priv_class = &aacenc_class,
820 };
float, planar
Definition: samplefmt.h:72
static const uint8_t aac_chan_configs[6][5]
default channel configurations
Definition: aacenc.c:138
void(* search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda)
Definition: aacenc.h:46
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sine window.
Definition: aac.h:160
This structure describes decoded (raw) audio or video data.
Definition: frame.h:135
static const uint8_t swb_size_1024_64[]
Definition: aacenc.c:64
int grouping[8]
window grouping (for e.g. AAC)
Definition: psymodel.h:67
AVOption.
Definition: opt.h:234
uint8_t ** bands
scalefactor band sizes for possible frame sizes
Definition: psymodel.h:82
Definition: aac.h:203
void(* mdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:94
static const AVClass aacenc_class
Definition: aacenc.c:799
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
Definition: kbdwin.c:26
#define SCALE_DIFF_ZERO
codebook index corresponding to zero scalefactor indices difference
Definition: aac.h:142
Definition: aac.h:56
Definition: aac.h:49
Definition: aac.h:50
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
Cleanup audio preprocessing module.
Definition: psymodel.c:130
int size
Definition: avcodec.h:974
struct AACEncContext::@3 buffer
AACCoefficientsEncoder * coder
Definition: aacenc.h:70
void avpriv_align_put_bits(PutBitContext *s)
Pad the bitstream with zeros up to the next byte boundary.
Definition: bitstream.c:45
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
Encode ics_info element.
Definition: aacenc.c:269
#define AAC_MAX_CHANNELS
Definition: aacenc.c:48
int common_window
Set if channels share a common 'IndividualChannelStream' in bitstream.
Definition: aac.h:249
av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens, const uint8_t **bands, const int *num_bands, int num_groups, const uint8_t *group_map)
Initialize psychoacoustic model.
Definition: psymodel.c:31
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
Definition: aac.h:251
static const uint8_t swb_size_128_8[]
Definition: aacenc.c:124
float lambda
Definition: aacenc.h:73
int profile
profile
Definition: avcodec.h:2622
AVCodec.
Definition: avcodec.h:2796
static const uint8_t swb_size_1024_8[]
Definition: aacenc.c:95
static const uint8_t swb_size_128_96[]
Definition: aacenc.c:108
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
Encode spectral coefficients processed by psychoacoustic model.
Definition: aacenc.c:418
int * num_bands
number of scalefactor bands for possible frame sizes
Definition: psymodel.h:83
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:198
const uint8_t ff_aac_num_swb_128[]
Definition: aactab.c:47
AACEncOptions options
encoding options
Definition: aacenc.h:56
AAC encoder context.
Definition: aacenc.h:54
uint8_t
#define av_cold
Definition: attributes.h:66
AVOptions.
#define WINDOW_FUNC(type)
Definition: aacenc.c:184
void(* quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size, int scale_idx, int cb, const float lambda)
Definition: aacenc.h:44
SingleChannelElement ch[2]
Definition: aac.h:253
int samplerate_index
MPEG-4 samplerate index.
Definition: aacenc.h:63
Definition: aac.h:52
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
const uint8_t * chan_map
channel configuration map
Definition: aacenc.h:65
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:74
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1164
const char * name
AudioFrameQueue afq
Definition: aacenc.h:74
static const uint8_t swb_size_1024_48[]
Definition: aacenc.c:70
uint8_t * data
Definition: avcodec.h:973
uint32_t tag
Definition: movenc.c:844
AVFloatDSPContext fdsp
Definition: aacenc.h:60
#define CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:658
int duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:991
static void adjust_frame_information(ChannelElement *cpe, int chans)
Produce integer coefficients from scalefactors provided by the model.
Definition: aacenc.c:304
static const AVOption aacenc_options[]
Definition: aacenc.c:791
const OptionDef options[]
Definition: avconv_opt.c:2187
static const uint8_t swb_size_1024_24[]
Definition: aacenc.c:83
float coeffs[1024]
coefficients for IMDCT
Definition: aac.h:236
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:123
#define CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: avcodec.h:713
static const int sizes[][2]
Definition: img2dec.c:46
#define AVERROR(e)
Definition: error.h:43
const uint8_t ff_aac_num_swb_1024[]
Definition: aactab.c:39
int last_frame
Definition: aacenc.h:72
sample_fmts
Definition: avconv_filter.c:68
#define CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:718
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:150
int stereo_mode
Definition: aacenc.h:34
g
Definition: yuv2rgb.c:535
float ff_aac_kbd_long_1024[1024]
Definition: aactab.c:36
int flags
CODEC_FLAG_*.
Definition: avcodec.h:1144
int amp[4]
Definition: aac.h:207
#define CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:611
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:168
const char * name
Name of the codec implementation.
Definition: avcodec.h:2803
int num_windows
number of windows in a frame
Definition: psymodel.h:66
static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
Definition: aacenc.c:483
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:158
static void put_bits(PutBitContext *s, int n, unsigned int value)
Write up to 31 bits into a bitstream.
Definition: put_bits.h:134
#define ff_mdct_init
Definition: fft.h:151
Definition: aac.h:55
int num_swb
number of scalefactor window bands
Definition: aac.h:166
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
#define FFMAX(a, b)
Definition: common.h:55
void(* search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s, SingleChannelElement *sce, const float lambda)
Definition: aacenc.h:40
static int put_bits_count(PutBitContext *s)
Definition: put_bits.h:67
#define FF_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:531
#define AACENC_FLAGS
Definition: aacenc.c:790
int bit_rate
the average bitrate
Definition: avcodec.h:1114
enum WindowSequence window_sequence[2]
Definition: aac.h:159
int cur_channel
Definition: aacenc.h:71
#define FFMIN(a, b)
Definition: common.h:57
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: aacenc.c:505
void(* analyze)(FFPsyContext *ctx, int channel, const float **coeffs, const FFPsyWindowInfo *wi)
Perform psychoacoustic analysis and set band info (threshold, energy) for a group of channels...
Definition: psymodel.h:122
#define FF_PROFILE_AAC_LOW
Definition: avcodec.h:2627
#define FF_PROFILE_UNKNOWN
Definition: avcodec.h:2623
static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS]
Table to remap channels from Libav's default order to AAC order.
Definition: aacenc.c:150
int pos[4]
Definition: aac.h:206
int channels
channel count
Definition: aacenc.h:64
int ff_alloc_packet(AVPacket *avpkt, int size)
Check AVPacket size and/or allocate data.
Definition: utils.c:1245
AAC definitions and structures.
FFTContext mdct128
short (128 samples) frame transform context
Definition: aacenc.h:59
PutBitContext pb
Definition: aacenc.h:57
static void(*const apply_window[4])(AVFloatDSPContext *fdsp, SingleChannelElement *sce, const float *audio)
Definition: aacenc.c:240
float ff_aac_pow34sf_tab[428]
Definition: aacenc.c:56
static const uint8_t swb_size_128_48[]
Definition: aacenc.c:112
static const uint8_t swb_size_128_24[]
Definition: aacenc.c:116
LIBAVUTIL_VERSION_INT
Definition: eval.c:55
if(ac->has_optimized_func)
static const uint8_t swb_size_1024_16[]
Definition: aacenc.c:89
static av_cold int aac_encode_end(AVCodecContext *avctx)
Definition: aacenc.c:674
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1811
static const uint8_t swb_size_1024_32[]
Definition: aacenc.c:77
NULL
Definition: eval.c:55
Libavcodec external API header.
AVSampleFormat
Audio Sample Formats.
Definition: samplefmt.h:61
static void put_audio_specific_config(AVCodecContext *avctx)
Make AAC audio config object.
Definition: aacenc.c:163
AV_SAMPLE_FMT_NONE
Definition: avconv_filter.c:68
int sample_rate
samples per second
Definition: avcodec.h:1791
float ff_aac_kbd_short_128[128]
Definition: aactab.c:37
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
Encode MS data.
Definition: aacenc.c:290
FFPsyWindowInfo(* window)(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type)
Suggest window sequence for channel.
Definition: psymodel.h:112
av_default_item_name
Definition: dnxhdenc.c:52
int frame_bits
number of bits used for the previously encoded frame
Definition: avcodec.h:2285
main external API structure.
Definition: avcodec.h:1050
static void(WINAPI *cond_broadcast)(pthread_cond_t *cond)
static void close(AVCodecParserContext *s)
Definition: h264_parser.c:490
int bits
number of bits used in the bitresevoir
Definition: psymodel.h:88
IndividualChannelStream ics
Definition: aac.h:228
void(* encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce, int win, int group_len, const float lambda)
Definition: aacenc.h:42
int extradata_size
Definition: avcodec.h:1165
uint8_t group_len[8]
Definition: aac.h:162
Describe the class of an AVClass context structure.
Definition: log.h:33
static void put_bitstream_info(AACEncContext *s, const char *name)
Write some auxiliary information about the created AAC file.
Definition: aacenc.c:462
static const uint8_t swb_size_1024_96[]
Definition: aacenc.c:58
int window_shape
window shape (sine/KBD/whatever)
Definition: psymodel.h:65
static void encode_pulses(AACEncContext *s, Pulse *pulse)
Encode pulse data.
Definition: aacenc.c:399
static const uint8_t swb_size_128_16[]
Definition: aacenc.c:120
const uint8_t * swb_sizes
table of scalefactor band sizes for a particular window
Definition: aac.h:165
FFPsyContext psy
Definition: aacenc.h:68
const uint32_t ff_aac_scalefactor_code[121]
Definition: aactab.c:55
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
Definition: aacenc.c:709
const struct FFPsyModel * model
encoder-specific model functions
Definition: psymodel.h:76
int ms_mode
Signals mid/side stereo flags coding mode (used by encoder)
Definition: aac.h:250
static const uint8_t * swb_size_1024[]
Definition: aacenc.c:101
struct FFPsyPreprocessContext * psypp
Definition: aacenc.h:69
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1130
uint8_t zeroes[128]
band is not coded (used by encoder)
Definition: aac.h:235
int sf_idx[128]
scalefactor indices (used by encoder)
Definition: aac.h:234
AVCodec ff_aac_encoder
Definition: aacenc.c:806
av_cold void avpriv_float_dsp_init(AVFloatDSPContext *fdsp, int bit_exact)
Initialize a float DSP context.
Definition: float_dsp.c:115
const int avpriv_mpeg4audio_sample_rates[16]
Definition: mpeg4audio.c:55
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_dlog(ac->avr,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
Y Spectral Band Replication.
Definition: mpeg4audio.h:64
int off
Definition: latmenc.c:31
float * samples
Definition: aacenc.h:79
static av_cold int aac_encode_init(AVCodecContext *avctx)
Definition: aacenc.c:724
common internal api header.
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:83
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:227
windowing related information
Definition: psymodel.h:63
#define ff_mdct_end
Definition: fft.h:152
av_cold struct FFPsyPreprocessContext * ff_psy_preprocess_init(AVCodecContext *avctx)
psychoacoustic model audio preprocessing initialization
Definition: psymodel.c:95
AACCoefficientsEncoder ff_aac_coders[]
Definition: aaccoder.c:1115
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
Preprocess several channel in audio frame in order to compress it better.
Definition: psymodel.c:118
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce)
Encode scalefactors.
Definition: aacenc.c:377
#define CODEC_CAP_EXPERIMENTAL
Codec is experimental and is thus avoided in favor of non experimental encoders.
Definition: avcodec.h:741
ChannelElement * cpe
channel elements
Definition: aacenc.h:67
Individual Channel Stream.
Definition: aac.h:157
float ff_aac_pow2sf_tab[428]
Definition: aac_tablegen.h:32
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:48
static av_cold int init(AVCodecParserContext *s)
Definition: h264_parser.c:499
static const uint8_t * swb_size_128[]
Definition: aacenc.c:128
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:247
void * priv_data
Definition: avcodec.h:1092
int start
Definition: aac.h:205
FFTContext mdct1024
long (1024 samples) frame transform context
Definition: aacenc.h:58
#define ERROR_IF(cond,...)
Definition: aacenc.c:50
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int *duration)
Remove frame(s) from the queue.
int channels
number of audio channels
Definition: avcodec.h:1792
int num_pulse
Definition: aac.h:204
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
Encode scalefactor band coding type.
Definition: aacenc.c:366
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
enum BandType band_type[128]
band types
Definition: aac.h:231
#define LIBAVCODEC_IDENT
Definition: version.h:43
int frame_number
Frame counter, set by libavcodec.
Definition: avcodec.h:1822
float ret_buf[2048]
PCM output buffer.
Definition: aac.h:238
void ff_aac_tableinit(void)
Definition: aac_tablegen.h:34
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, int common_window)
Encode one channel of audio data.
Definition: aacenc.c:443
static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, float *audio)
Definition: aacenc.c:249
struct FFPsyContext::@56 bitres
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:169
AAC data declarations.
av_cold void ff_psy_end(FFPsyContext *ctx)
Cleanup model context at the end.
Definition: psymodel.c:76
This structure stores compressed data.
Definition: avcodec.h:950
int delay
Codec delay.
Definition: avcodec.h:1212
int window_type[3]
window type (short/long/transitional, etc.) - current, previous and next
Definition: psymodel.h:64
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:179
static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
Definition: aacenc.c:689
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:966
void ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
#define FF_ALLOCZ_OR_GOTO(ctx, p, size, label)
Definition: internal.h:126
float * planar_samples[6]
saved preprocessed input
Definition: aacenc.h:61
bitstream writer API